Clicking on apply causes pjsip/sip-tls endpoints to drop

Distro Discussion & Help
41

Tony

retrieve_conf grabs all the settings in the MySQL DB for everything you have in the FreePBX GUI. It re-writes the asterisk config files.

core reload takes all the dialplan in asterisk and reloads it into the running config.

If you do a core reload and nothing breaks I can see why you think its a FreePBX issue but if you do a retreive_conf without changing a thing in the GUI it takes and writes out the same configs that you currently have and do a core reload it breaks.

You have confirmed that nothing changes in the /etc/asterisk directory when you diff before and after retrieve_conf without making a change in the GUI yet a core reload now breaks it. If nothing is truly changing in the configs its clear Asterisk must be breaking because the conf file has a new time stamp or something weird.

A simple test would be to just touch those files so the time and date stamps get updated and do a core reload and see what happens. My guess is it will break. Take that data back to Digium and hopefully they can get it fixed up.

42

Tony, I was thinking along the same lines as your ā€˜touchā€™ idea, i am going to try that tonight.

thanks

43

ok, so here are my results.

first, i did a core reload and there were no dropped connections
next, i made a copy of /etc/asterisk and ran a diff to verify
next, i touch /etc/asterisk/*
run diff again, no change in files

now, i run core reload ā€¦and i drop connectionsā€¦so, i guess that completely rules out freepbx
although, i still suspect that andrew is the cigarette smoking man !!!

anyway, i am off to asterisk community, if I have some time over the weekend, i will try to touch files one by one to see if it is a specific file

thanks again for all your help

1 Like
44

I would start with your pjsip files as I bet those are the ones.

45

I am not finding anything on the asterisk issue tracker. Maybe @malcolmd might have some ideas.

46

tony, thanks for the suggestions, how do i get a hold of @malcomd, through the community or irc?

47

I already pinged him here to see if he had any ideas.

48

Thereā€™s this:
https://issues.asterisk.org/jira/browse/ASTERISK-27001

and this:
https://issues.asterisk.org/jira/browse/ASTERISK-27147

those are the only two things in the hopper right now that come to mind that are in the same general area.

49

thanks again

50

tony, just got this back on my asterisk query, do you know anything about this limitation or the ā€˜allow_reloadā€™ option?

51

That setting is force enabled. You canā€™t turn it off. We could expose itā€¦

If youā€™d like you can open a ticket up on freepbx and we can expose that setting.

52

Yep, itā€™s an Asterisk bug allright. For those that didnā€™t follow the thread on asterisk.org, hereā€™s the ticket.

https://issues.asterisk.org/jira/browse/ASTERISK-27148

53

andrew, yes, thanks but can you point me to some docs about this, right now, i donā€™t understand enough to know how I should set it.

thanks

https://issues.freepbx.org/browse/FREEPBX-15421

54

You canā€™t change the setting at this time.

55

yes, what i meant was, after it is exposed, i am not sure which way to set, thats why i would like to read up on it.

56

Ohh this discussion is answering my questions why all Phones do not work anymore after apply. :smiley:

All phones can make an ECHO Test but calling no other phone.

All Sangoma Phones have no sound anymore. But also not after a reboot of FreePBX. Maybe a configuration error in EPM.

57

Why would you think this is a EPM bug. Itā€™s clearly a asterisk bug

58

Until itā€™s exposed how could we tell you how to do it. You are asking for us to tell you how to do something that doesnā€™t exist in FreePBX at this time to do.

1 Like
59

I meaned a confuguration error on my side. :joy::joy::joy:

My test Environment uses tls srtp and is an external device. Basefile settings needet to be changed for this case :slight_smile:

60

andrew, tony,

could you please take a look at the comment made by george, i can try to reproduce what he is asking, but it will take time. if you have any input, that would be greatly appreciated.

https://issues.asterisk.org/jira/browse/ASTERISK-27148